1. Field of the Invention
The invention is generally related to an equalization system that improves the sound quality of an audio system in a listening room. In particular, the invention relates to an equalization system that improves the sound quality of an audio system based upon near- and far-field measurement data.
2. Related Art
The aim of a high-quality audio system is to faithfully reproduce a recorded acoustic event, such as a concert hall experience, in smaller enclosed spaces, such as a listening room, a home theater or entertainment center, a PC environment, or an automobile.
The perceived sound quality of an audio system in smaller enclosed spaces depends on several factors: quality and radiation characteristics of the loudspeakers (e.g., on- and off-axis frequency responses); placement of the loudspeakers at their connect positions according to the standard (for example, ITU 5.1/7.1); acoustics of the room in general (low frequency modes, reverb time, frequency-dependent absorption, effects of room geometry and dimensions, location of furniture, etc.); and nearby reflective surfaces and obstacles (e.g., on-wall mounting, bookshelves, TV sets, etc.).
In order to provide an optimum listening experience in such enclosed spaces, a digital “room equalization” system may be used. In general, equalization is the process of either boosting or attenuating certain frequency components in a signal. There are several types of equalization, each with a different pattern of attenuation or boost. Examples are a high-pass filter, bandpass filter, graphic equalizer, and parametric equalizer.
In a multiband parametric equalizer (“EQ”), center frequency, bandwidth (Q-factor) or peak shape, and gain (peak amplitude above a given reference) in each of the bands may be adjusted to flatten a measured frequency response at a listening location (e.g., a seat in a listening room), Typically, a cascade of second-order IIR (“infinite impulse response”) filter sections (“biquads”) is used to control frequency response. A digital signal processor (“DSP”) may generate test signals for each loudspeaker (e.g., either white or pink noise or logarithmic sweeps), in order to capture room responses at a desired listening location. For that purpose, an omni-directional microphone may be positioned at the listening location and connected to a signal analyzer or back to the DSP.
In FIG. 1, a test system 100 that uses an equalizer to produce a signal at the listening location that resembles the input signal is shown. In an example of operation, signal source 104 produces a test signal, which is amplified by the preamplifier 106 and processed by the equalizer 108. The test signal is then amplified by the power amplifier 110 and transmitted to a loudspeaker 112. The loudspeaker 112 reproduces the test signal as an acoustic pressure wave that is emitted from the loudspeaker 112, which is then picked up by the test microphone 116 and passed to a signal analyzer 120.
In this example of operation, the received test signal is observed at the signal analyzer 120 and, in response, the test signal may be adjusted accordingly through the equalizer 108. In other implementations, the test microphone 116 may be directly in signal communication with the equalizer 108, where the received test signal may be automatically processed by the equalizer 108, which may include digital signal processors (“DSPs”). Additionally, the test microphone 116 may be positioned at a listening location in a room or hall, where it can then capture the impulse responses at that particular listening location.
In this example, if the equalizer 108 is a parametric EQ with multiple filters, the multiple filters may be set manually, so that, for example, a displayed response curve, on an output device (not shown) in signal communication with the equalizer 108, becomes smoother, or automatically, with the aid of an external processor such as, for example a personal computer (“PC”) or design logic built into the DSP itself. In general, it is difficult and suboptimal to adjust a set of cascaded parametric filter sections because of overlap. Two or more of the parametric filter sections may affect the same frequency band of interest, which leads to the difficulty that a large number of parameters need to be adjusted simultaneously. At low frequencies, it is important to accurately suppress individual room modes. In order to avoid approximation errors and quantization noise, a FIR (“finite impulse response”) filter may be directly used and operated at a low sample rate (for example, utilizing decimation) to minimize processing cost.
In adjusting a frequency response, it is important to distinguish between resonances (e.g., loudspeaker cabinet material resonances, or standing waves at low frequencies in rooms) and interferences due to multiple reflections that lead to nulls (dips) in the frequency response. Resonances and room modes need to be suppressed, e.g., with a notch filter, while narrow-band interference dips strongly depend on the measurement position and generally should be left unaltered. An attempt to correct narrow-band interference dips may introduce high-gain peak filters that are perceived as resonances.
In an intermediate frequency band (between approximately 100 Hz to 1000 Hz), it is desirable to correct errors related to the source only, not the whole listening room. For example, eliminating sonic differences between the main stereo speakers and the center speaker, which may be close to a reflective surface such as a TV set, leads to an improved stereo image. This so-called “source-related” correction is independent of a particular listening location, whereas a complete room correction would be valid at a single point only.
At high frequencies (i.e., greater than 1000 Hz), the in-room response is normally not flat, but decreases with frequency. This may be addressed by a so-called “target function.” Equalization is performed such that the final response approximates the target function. However, the correct target function choice depends on the absorption properties of the particular room and the radiation characteristics of the loudspeakers, and is thus a priori unknown. In a (domestic) listening room solution, a set of near-field measurements close to the loudspeakers provides frequency response data above typically 1000 Hz, thus eliminating the need for a target function. In all automobile, an adjustable target function may be provided with the EQ algorithm.
Along with the foregoing considerations, there are many other factors to be considered when trying to optimize the sound quality audio systems utilized in small spaces such as listening rooms or cars. Therefore, there is always a continuing need to improve the sound quality of these audio systems, in particular, by improving the fully-automated equalization of the responses of loudspeakers located in these small spaces.